similar to: FW: MeetMe Conferencing and Marked Mode

I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks _____________________ Kevin Savoy Business Unit Telecom Analyst 2218 4th

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

I have discovered an issue on my system after upgrading from 1.2.13 to 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I have confirmed this on multiple phones. When the called person answers and tries to transfer the call to another extension, the call successfully transfers, however the person answering the transfer cannot hear the person that called in, the caller. My

I am having an issue with 1.4 where we can't successfully transfer a call directly to a voicemail box. We hit "Transfer" on the phone and dial the mailbox number we want to send it to, My dial plan for this is: exten=>_*40XX,n,Voicemail(${EXTEN:1},u) The voicemail system picks up and starts to play its message and at this point. We should then hit "Transfer"

We are looking at using Asterisk as a call recording server for an Avaya VoIP S8700 system in a multi-site VoIP Call Center. All calls will be coming in to one location and sent out via VoIP to other call centers. What kind of specs should we be looking at purchasing for our Asterisk server to be record up 200-300 calls simultaneously? Linux runs in 64 bit architecture, but does Asterisk

So no one else is having issues with MySQL and 1.4? I'm the only one? -----Original Message----- From: Savoy, Kevin - Williston, ND Sent: Wednesday, December 27, 2006 2:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] cdr_addon_mysql.so did not register itself duringload Well the addons from 1.4 are installed. This original Asterisk

VMukti.com ----- Original Message ----- From: "Anselm Martin Hoffmeister" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Filesharing + video + voice supported Soft phone Date: Mon, 17 Sep 2007 15:06:03 +0200 Am Montag, den 17.09.2007, 05:09 -0700 schrieb satish patel: > Dear all > > I have setup of

VMukti.com ----- Original Message ----- From: "Goltsios Theodore" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Filesharing + video + voice supported Soft phone Date: Mon, 17 Sep 2007 16:57:25 +0300 There is for sure X-lite and other similar but you won't get file sharing which is meaningless either way. If

Ok so something is missing. I get the below for those two lines. checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... no I even installed the mysql-devel as Bradley Watkins suggested and still it says no. What do I need to make that say yes? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com

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Sorry if this is a repeat but I didn't receive a copy of it so I'm not sure it actually posted. The below worked for normal transfers. Now here is another situation. When we try to transfer a call directly to voicemail it plays the voicemail message but we can't transfer the call. The only way I could get it to work was to do a conference and then drop out of that conference. My

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This worked. Great and thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Carlos Chavez Sent: Wednesday, February 07, 2007 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly On Wed, 2007-02-07 at 14:12

Well the addons from 1.4 are installed. This original Asterisk 1.2.x box was created by my predecessor and he had the cdr_addon_mysql.so and res_config_mysql.so files on a server that we copied to any new installation. I'm not sure where he got these files. As far as I can tell shouldn't the install of the addons create these files? If not where do I get them from? I've done a search

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Savoy, Kevin - Williston, ND > Sent: Thursday, December 28, 2006 12:20 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: FW: [asterisk-users] cdr_addon_mysql.so did not > register itselfduringload >

Did you install the MySQL libraries? Debian's command is - apt-get install libmysqlclient15-dev Andy -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthias Urlichs Sent: 27 November 2008 16:05 To: asterisk-users at lists.digium.com Subject: [SPAM] - Re: [asterisk-users] FW: cdr_addon_mysql.so did

Thanks for the feedback. I did some research and it looks like you were calling over international lines. It also appears that there was high than average static on the line, which is not normal for my system. It's true that I threw my recordings together quickly and the beep was supposed to be funny - it was actually me saying "beeeeeep". However, the static and noise you received

I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not

I've been asked by my company to setup a conferencing system to support up to 400 people on a conference calls, where all users will be dialling in frpm the PSTN. I am exploring using Asterisk meetme to do this. I have two questions in relation to this:- For Meetme conferences is it better to have all participants to dial in via SIP provider terminating to Asterisk via SIP/IAX, or use

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this information in order to be able to bill. As teleconferencing is the only application of the